The invention relates to a signal processing method and apparatus for block-coded audio signals of a communication system.
Transmitting and receiving devices are used for message processing and transmission in communications systems having a message transmission path between a message origin and a message destination. The message produced by the message origin is transmitted by the transmitting device via a message channel to the receiving device, which subsequently emits the received message to the message destination. The message processing and transmission can in this case be carried out in a preferred transmission direction or in both transmission directions (duplex operation).
"Message" is a generic term which represents both the meaning content (information) and the physical representation (signal). Signals may represent, for example,
(1) pictures, PA0 (2) spoken words, PA0 (3) written words, PA0 (4) encrypted words or pictures. PA0 (1) A. Papoulis: "A new algorithm in Spectral Analysis and Band-Limited Extrapolation"; IEEE Transactions on Circuits and Systems, Volume 22 (9), pages 735 ff., 1975 and PA0 (2) R. Sottek: "Modelle zur Signalverarbeitung im menschlichen Gehor" [Models for signal processing in human hearing]; Thesis at the Institute for Electrical Telecommunications, RWTH Aachen 1993.
The type of transmission according to (1), (2) and (3) is in this case normally characterized by continuous (analog) signals, while non-continuous signals (for example pulses, digital signals) are normally produced for the type of transmission according to (4).
The present invention primarily relates to the transmission of audio messages (for example voice or music messages, etc.). However, it can also be applied to other messages, such as appropriately processed video messages, for example.
Either continuous signals (pure analog signals) or a mixture of continuous and non-continuous signals occur as possible signal forms for an audio communications system, using A/D (analog to digital) converters and D/A (digital to analog) converters. Devices which are specific to the message type are in each case required for the functions of "transmitting" and "receiving". The question as to which of these devices is finally used also depends, inter alia, on the communications channel which is used as the basis for the audio communications system. The present invention thus primarily relates to telecommunications systems, which have a wire-free telecommunications channel. Telecommunications systems having such a structure are, for example, cordless telephones to the DECT standard (Digital European Cordless Telecommunication; cf. (1) European Telecommunication Standard; prETS 300 175-1 . . . 9, October 1992, Parts 1 to 9, ETS-Institute 06921 Sofia Antipoles, France; (2) Nachrichtentechnik Elektronik 42 (Telecommunications Electronics 42) (January/February 1992), No. 1, Berlin; U. Pilger: "Struktur des DECT-Standards" (Structure of the DECT Standard); pages 23 to 29; (3) Philips Telecommunication Review: "DECT, Universal Cordless Access System"; Vol. 49, No. 3, September 1991, pages 68 to 73) or mobile radio telephones to the GSM standard (Groupe Speciale Mobile Systems for Mobile Communication; cf. Informatik Spektrum (Information Spectrum), Springer Press Berlin, Year 14, 1991, No. 3, pages 137 to 152, "Der GSM-Standard--Grundlage fur digitale europaische Mobilfunknetze" (The GSM Standard Basis for digital European mobile radio networks)).
The DECT cordless telephone and the GSM mobile radio telephone are audio communications systems in which block-coded audio signals--for example signals which are coded using the TDMA or CDMA method (Time Division Multiple Access or Code Division Multiple Access)--are processed. The message transmitted using these telephones as a rule comprises, according to the above definition of message types, a mixture of continuous and non-continuous signals. This signal mixture is in this case produced by the use of analog/digital and digital/analog converters.
FIG. 1 shows a DECT cordless telephone having a cordless base station FT (Fixed Termination) to which a maximum of twelve cordless mobile sections (PT1 . . . PT12 (Portable Termination) are assigned for cordless telecommunication via a radio channel FK. Cordless base stations designed in such a way have been introduced to the market using the product name "Gigaset 952"--cf. DE-Z: the German journal Funkschau December 1993, pages 24 and 25; "Digitale Freiheit--Gigaset 952: Das erste DECT-Telefon" (Digital freedom--Gigaset 952: The first DECT telephone); author: G. Weckwerth--1993. This design was essentially also known before this from DE-Z: the German journal Funkschau October 1993; pages 74 to 77; title: "Digital kommunizieren mit DECT-DECT-Chipsatz von Philips" (Communicate digitally using the DECT-DECT chip set from Philips); author: Dr. J. Nieder and WO 94/10812 (FIG. 1 with the associated description).
FIG. 2 shows the principle of the design of the DECT-specific cordless mobile section PT1 . . . PT12, as is used for the transmission of voice messages in the cordless telephone. Cordless mobile sections designed in such a way have likewise been introduced to the market using the product name "Gigaset 952"--cf. DE-Z: the German journal Funkschau December 1993, pages 24 and 25; "Digitale Freiheit--Gigaset 952: Das erste DECT-Telefon" (Digital freedom--Gigaset 952: The first DECT telephone); author: G. Weckwerth--1993. This design was essentially also known before this from DE-Z: the German journal Funkschau October 1993; pages 74 to 77; title: "Digital kommunizieren mit DECT-DECT-Chipsatz von Philips" (Communicate digitally using the DECT-DECT chip set from Philips); author: Dr. J. Nieder and WO 94/10812 (FIG. 1 with the associated description).
Block-oriented coding methods (for example TDMA methods) are used for the transmission of voice and/or music signals (audio signals) with the DECT cordless telephone, in order on the one hand to use a correlation between signal sections which follow one another in time for data reduction and/or, on the other hand, to carry out block-oriented error protection by means of parity bits. If the transmission of the digitally coded signals is disturbed, then bit errors obviously occur which can be compensated for, if the error rate is low, by the redundancy mechanisms which are assigned to the digitally coded signal. However, if the bit error rates reach higher levels, an error correction is no longer possible and an entire signal block will in consequence be identified as being faulty and will be rejected. There are a number of options at the receiver end for coping with such signal blocks which have been identified as being faulty and have been rejected.
A first option, which is disclosed in WO 94/10769, comprises "squelching" the appropriate signal block which has been identified as being faulty, that is to say changing the code in an appropriate manner, for example by means of a sequence of zeros. This method is now used in digital DECT cordless telephones, such as Gigaset 952.
A second option for error correction is to assume that the error which has occurred is only minor. However, in this case, it is necessary to distinguish whether the algorithm can be used to identify the importance of the respectively disturbed bits. In the case of conventional linear coding, for example, a disturbed less significant bit (LSB=Least Significant Bit) would scarcely produce any audible errors, while an incorrectly set more significant bit (MSB=Most Significant Bit) would produce severe sudden changes in the transmission signal and thus crackling-like interference. Meanwhile, it is not possible in all cases to identify directly how severe the specific interference to be expected will be.
An entirely different way to correct errors in disturbed audio signals is proposed in the documents:
A method is known from each of the cited documents, in which signal errors in the audio signal which are caused by interference are masked by interpolation of the signal. The disadvantages in the case of this method are, on the one hand, the high technical complexity which, under some circumstances, demands the complete computation power of a currently marketed digital signal processor (DSP=Digital Signal Processor) and, on the other hand, makes the algorithmic delay of the signal necessary, if processing is carried out in the frequency domain using Fourier transformation. This delay would not be tolerable, for example, in the case of telephony, particularly cordless telephony.
A method for the transmission of digital audio signals is disclosed in German reference DE-41 11 131 A1, in which a substitution signal which is correlated with the signal is generated and buffer-stored for processing of the signals, at least one first incorrectly transmitted signal section is determined in the signal, the first signal section of the signal is replaced by the substitution signal, and substitution-dependent artefacts in the signal are suppressed.